--- 1/draft-ietf-rtcweb-rtp-usage-21.txt 2015-02-09 04:16:22.232815287 -0800 +++ 2/draft-ietf-rtcweb-rtp-usage-22.txt 2015-02-09 04:16:22.340817899 -0800 @@ -1,21 +1,21 @@ RTCWEB Working Group C. S. Perkins Internet-Draft University of Glasgow Intended status: Standards Track M. Westerlund -Expires: May 30, 2015 Ericsson +Expires: August 13, 2015 Ericsson J. Ott Aalto University - November 26, 2014 + February 09, 2015 Web Real-Time Communication (WebRTC): Media Transport and Use of RTP - draft-ietf-rtcweb-rtp-usage-21 + draft-ietf-rtcweb-rtp-usage-22 Abstract The Web Real-Time Communication (WebRTC) framework provides support for direct interactive rich communication using audio, video, text, collaboration, games, etc. between two peers' web-browsers. This memo describes the media transport aspects of the WebRTC framework. It specifies how the Real-time Transport Protocol (RTP) is used in the WebRTC context, and gives requirements for which RTP features, profiles, and extensions need to be supported. @@ -28,25 +28,25 @@ Internet-Drafts are working documents of the Internet Engineering Task Force (IETF). Note that other groups may also distribute working documents as Internet-Drafts. The list of current Internet- Drafts is at http://datatracker.ietf.org/drafts/current/. Internet-Drafts are draft documents valid for a maximum of six months and may be updated, replaced, or obsoleted by other documents at any time. It is inappropriate to use Internet-Drafts as reference material or to cite them other than as "work in progress." - This Internet-Draft will expire on May 30, 2015. + This Internet-Draft will expire on August 13, 2015. Copyright Notice - Copyright (c) 2014 IETF Trust and the persons identified as the + Copyright (c) 2015 IETF Trust and the persons identified as the document authors. All rights reserved. This document is subject to BCP 78 and the IETF Trust's Legal Provisions Relating to IETF Documents (http://trustee.ietf.org/license-info) in effect on the date of publication of this document. Please review these documents carefully, as they describe your rights and restrictions with respect to this document. Code Components extracted from this document must include Simplified BSD License text as described in Section 4.e of the Trust Legal Provisions and are provided without warranty as @@ -915,21 +915,21 @@ The use of Forward Error Correction (FEC) can provide an effective protection against some degree of packet loss, at the cost of steady bandwidth overhead. There are several FEC schemes that are defined for use with RTP. Some of these schemes are specific to a particular RTP payload format, others operate across RTP packets and can be used with any payload format. It needs to be noted that using redundant encoding or FEC will lead to increased play out delay, which needs to be considered when choosing FEC schemes and their parameters. WebRTC endpoints MUST follow the recommendations for FEC use given in - [I-D.uberti-rtcweb-fec]. WebRTC endpoints MAY support other types of + [I-D.ietf-rtcweb-fec]. WebRTC endpoints MAY support other types of FEC, but these MUST be negotiated before they are used. 7. WebRTC Use of RTP: Rate Control and Media Adaptation WebRTC will be used in heterogeneous network environments using a variety set of link technologies, including both wired and wireless links, to interconnect potentially large groups of users around the world. As a result, the network paths between users can have widely varying one-way delays, available bit-rates, load levels, and traffic mixtures. Individual end-points can send one or more RTP packet @@ -1826,64 +1826,64 @@ [I-D.ietf-avtcore-multi-media-rtp-session] Westerlund, M., Perkins, C., and J. Lennox, "Sending Multiple Types of Media in a Single RTP Session", draft- ietf-avtcore-multi-media-rtp-session-06 (work in progress), October 2014. [I-D.ietf-avtcore-rtp-circuit-breakers] Perkins, C. and V. Singh, "Multimedia Congestion Control: Circuit Breakers for Unicast RTP Sessions", draft-ietf- - avtcore-rtp-circuit-breakers-07 (work in progress), - October 2014. + avtcore-rtp-circuit-breakers-08 (work in progress), + December 2014. [I-D.ietf-avtcore-rtp-multi-stream-optimisation] Lennox, J., Westerlund, M., Wu, Q., and C. Perkins, "Sending Multiple Media Streams in a Single RTP Session: Grouping RTCP Reception Statistics and Other Feedback ", draft-ietf-avtcore-rtp-multi-stream-optimisation-00 (work in progress), July 2013. [I-D.ietf-avtcore-rtp-multi-stream] Lennox, J., Westerlund, M., Wu, W., and C. Perkins, "Sending Multiple Media Streams in a Single RTP Session", draft-ietf-avtcore-rtp-multi-stream-06 (work in progress), October 2014. [I-D.ietf-mmusic-sdp-bundle-negotiation] Holmberg, C., Alvestrand, H., and C. Jennings, "Negotiating Media Multiplexing Using the Session Description Protocol (SDP)", draft-ietf-mmusic-sdp-bundle- - negotiation-12 (work in progress), October 2014. + negotiation-16 (work in progress), January 2015. [I-D.ietf-rtcweb-audio] Valin, J. and C. Bran, "WebRTC Audio Codec and Processing Requirements", draft-ietf-rtcweb-audio-07 (work in progress), October 2014. + [I-D.ietf-rtcweb-fec] + Uberti, J., "WebRTC Forward Error Correction + Requirements", draft-ietf-rtcweb-fec-00 (work in + progress), February 2015. + [I-D.ietf-rtcweb-security-arch] Rescorla, E., "WebRTC Security Architecture", draft-ietf- rtcweb-security-arch-10 (work in progress), July 2014. [I-D.ietf-rtcweb-security] Rescorla, E., "Security Considerations for WebRTC", draft- ietf-rtcweb-security-07 (work in progress), July 2014. [I-D.ietf-rtcweb-video] Roach, A., "WebRTC Video Processing and Codec - Requirements", draft-ietf-rtcweb-video-02 (work in - progress), October 2014. - - [I-D.uberti-rtcweb-fec] - Uberti, J., "WebRTC Forward Error Correction - Requirements", draft-uberti-rtcweb-fec-00 (work in - progress), October 2014. + Requirements", draft-ietf-rtcweb-video-03 (work in + progress), November 2014. [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate Requirement Levels", BCP 14, RFC 2119, March 1997. [RFC2736] Handley, M. and C. Perkins, "Guidelines for Writers of RTP Payload Format Specifications", BCP 36, RFC 2736, December 1999. [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. Jacobson, "RTP: A Transport Protocol for Real-Time @@ -1982,54 +1982,53 @@ [I-D.ietf-avtcore-rtp-topologies-update] Westerlund, M. and S. Wenger, "RTP Topologies", draft- ietf-avtcore-rtp-topologies-update-05 (work in progress), November 2014. [I-D.ietf-avtext-rtp-grouping-taxonomy] Lennox, J., Gross, K., Nandakumar, S., and G. Salgueiro, "A Taxonomy of Grouping Semantics and Mechanisms for Real- Time Transport Protocol (RTP) Sources", draft-ietf-avtext- - rtp-grouping-taxonomy-03 (work in progress), November - 2014. + rtp-grouping-taxonomy-05 (work in progress), January 2015. [I-D.ietf-mmusic-msid] Alvestrand, H., "WebRTC MediaStream Identification in the Session Description Protocol", draft-ietf-mmusic-msid-07 (work in progress), October 2014. [I-D.ietf-mmusic-sdp-bundle-negotiation] Holmberg, C., Alvestrand, H., and C. Jennings, "Negotiating Media Multiplexing Using the Session Description Protocol (SDP)", draft-ietf-mmusic-sdp-bundle- - negotiation-12 (work in progress), October 2014. + negotiation-16 (work in progress), January 2015. [I-D.ietf-payload-rtp-howto] Westerlund, M., "How to Write an RTP Payload Format", draft-ietf-payload-rtp-howto-13 (work in progress), January 2014. [I-D.ietf-rmcat-cc-requirements] Jesup, R. and Z. Sarker, "Congestion Control Requirements for Interactive Real-Time Media", draft-ietf-rmcat-cc- - requirements-08 (work in progress), November 2014. + requirements-09 (work in progress), December 2014. [I-D.ietf-rtcweb-overview] Alvestrand, H., "Overview: Real Time Protocols for - Browser-based Applications", draft-ietf-rtcweb-overview-12 - (work in progress), October 2014. + Browser-based Applications", draft-ietf-rtcweb-overview-13 + (work in progress), November 2014. [I-D.ietf-rtcweb-use-cases-and-requirements] Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real- Time Communication Use-cases and Requirements", draft- - ietf-rtcweb-use-cases-and-requirements-14 (work in - progress), February 2014. + ietf-rtcweb-use-cases-and-requirements-16 (work in + progress), January 2015. [I-D.ietf-tsvwg-rtcweb-qos] Dhesikan, S., Jennings, C., Druta, D., Jones, P., and J. Polk, "DSCP and other packet markings for RTCWeb QoS", draft-ietf-tsvwg-rtcweb-qos-03 (work in progress), November 2014. [RFC3611] Friedman, T., Caceres, R., and A. Clark, "RTP Control Protocol Extended Reports (RTCP XR)", RFC 3611, November 2003.